callers type 1 but no reaction ...

fabianus

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#1
Hello all !


I created an IVR and use it as a break-out IVR from a queue. But some clients complain that when they type 1 (which is to break-out and leave a message on the AM) nothing happens. Anybody could help to get this fixed ?

Thanks a lot for any feedback !

Regards,
Fabianus
 

ramoncio

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#2
This looks as a dmtf problem.
Ho do the calls come in? SIP, PRI, BRI, E1, analog lines?
 

fabianus

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#3
Hi ramoncio,

thanks for your concern !

The calls come in by sip.

Regards,
Fabianus
 

ramoncio

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#4
What codec do you use?
You can see it with "sip show channels" from your asterisk console.
If you use g711a/u you can try inbound or rfc2833 dmtf modes in your peer config but if you use higher compression codecs, as g729 you can't use inband and must use rfc2833 in your configs.
 

fabianus

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#5
Hi Ramoncio,

I use the g729 codec. So I suppose that I have to user the rfc2833 dmtf mode.
Now, as I do not understand a lot about this ... ;-) could you be so kind and let me know where I should set this ?

Thanks for your kind support !

Regards,
Fabianus
 

dicko

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#6
Further (depending on the version of g729, "digium" or "otherwise" that you are using) you will need enough licenses to allow the g729 signalling to pass to the IVR. Asterisk is a "back to back" user agent, as such g729 devices will be "passed through" from g729 to g729 device (VSP to handset/softphone IF they have the license) but unless the asterisk system itself has a license for g729, my understanding is that the trans-code will be denied (at least by the digium codec). $10 dollars a (concurrent user) license, from www.digium.com
 

fabianus

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#7

dicko

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#8
Then I guess you are at the mercy of the Latvians, B)

check /var/log/asterisk/full for what is maybe not working.
 

fabianus

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#9
hi dicko,

why do you think that this is a limitation coming from the g729 ?
I think ramoncio is right, and I have to set the right dmtf mode. The problem is that I don't know how/where to do this ...

Regards,
Fabianus
 

dicko

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#10
hehe, Ramancio is ALWAYS right, you can set it in your sip trunks->incoming settings, but I believe unless you changed it that will be the default dtmfmode, if you change it to inband or info then the codec must be capable of handling that, I am not authoritative as to whether the codec you are using can do that, (I know the digium codec (with an appropriate license) can. (with limited success))
 

ramoncio

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#11
First read through this:
http://www.voip-info.org/wiki/view/Asterisk+DTMF

Then enable dtmf debug in logger.conf. This will help you debug the problem.
Maybe SIP INFO is the recommended dtmf mode for your sip provider. You can also ask their technical support.
 

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